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import os, math, torch, cv2
from PIL import Image
from omegaconf import OmegaConf
from tqdm import tqdm
from diffusers import AutoencoderKLTemporalDecoder
from diffusers.schedulers import EulerDiscreteScheduler
from transformers import WhisperModel, CLIPVisionModelWithProjection, AutoFeatureExtractor
from src.utils.util import save_videos_grid, seed_everything
from src.dataset.test_preprocess import process_bbox, image_audio_to_tensor
from src.models.base.unet_spatio_temporal_condition import UNetSpatioTemporalConditionModel, add_ip_adapters
from src.pipelines.pipeline_sonic import SonicPipeline
from src.models.audio_adapter.audio_proj import AudioProjModel
from src.models.audio_adapter.audio_to_bucket import Audio2bucketModel
from src.utils.RIFE.RIFE_HDv3 import RIFEModel
from src.dataset.face_align.align import AlignImage
BASE_DIR = os.path.dirname(os.path.abspath(__file__))
# ------------------------------------------------------------------
# single image + speech → video-tensor generator
# ------------------------------------------------------------------
def test(pipe, cfg, wav_enc, audio_pe, audio2bucket, image_encoder,
width, height, batch):
# -------- 배치 차원 정리 ---------------------------------------------
for k, v in batch.items():
if isinstance(v, torch.Tensor):
batch[k] = v.unsqueeze(0).to(pipe.device).float()
ref_img = batch["ref_img"] # (1,C,H,W)
clip_img = batch["clip_images"]
face_mask = batch["face_mask"]
image_embeds = image_encoder(clip_img).image_embeds
audio_feature = batch["audio_feature"] # (1,80,T)
audio_len = int(batch["audio_len"])
step = max(1, int(cfg.step)) # 최소 1 보장
# -------- Whisper 인코딩 --------------------------------------------
window = 16_000 # 1-초 단위
audio_prompts, last_prompts = [], []
for i in range(0, audio_feature.shape[-1], window):
chunk = audio_feature[:, :, i:i+window]
hs_all = wav_enc.encoder(chunk, output_hidden_states=True).hidden_states
last_hid = wav_enc.encoder(chunk).last_hidden_state.unsqueeze(-2) # (1,t,1,384)
audio_prompts.append(torch.stack(hs_all, dim=2)) # (1,t,12,384)
last_prompts.append(last_hid) # (1,t,1,384)
if not audio_prompts:
raise ValueError("[ERROR] No speech recognised in the provided audio.")
audio_prompts = torch.cat(audio_prompts, dim=1) # (1,T,12,384)
last_prompts = torch.cat(last_prompts, dim=1) # (1,T,1,384)
# -------- 앞뒤 padding ----------------------------------------------
audio_prompts = torch.cat(
[torch.zeros_like(audio_prompts[:, :4]),
audio_prompts,
torch.zeros_like(audio_prompts[:, :6])], dim=1)
last_prompts = torch.cat(
[torch.zeros_like(last_prompts[:, :24]),
last_prompts,
torch.zeros_like(last_prompts[:, :26])], dim=1)
total_tokens = audio_prompts.shape[1]
num_chunks = max(1, math.ceil(total_tokens / (2 * step)))
ref_list, audio_list, uncond_list, motion_buckets = [], [], [], []
for i in tqdm(range(num_chunks)):
start = i * 2 * step
# ------------------------------------------------------------
# cond_clip : (bz, f=1, w=10, b=5, c=384)
# bucket_clip: (bz, f=1, w=50, b=1, c=384)
# Whisper-tiny 는 hidden_state 층 수가 2 → 5 로 패딩
# ------------------------------------------------------------
clip_raw = audio_prompts[:, start:start + 10] # (1, ≤10, L, 384)
if clip_raw.shape[1] < 10: # w 패딩
pad_w = torch.zeros_like(clip_raw[:, :10 - clip_raw.shape[1]])
clip_raw = torch.cat([clip_raw, pad_w], dim=1)
# ---- L(=layers) 패딩: 부족하면 마지막 layer 를 반복 ----------
L_now = clip_raw.shape[2]
if L_now < 5:
pad_L = clip_raw[:, :, -1:].repeat(1, 1, 5 - L_now, 1)
clip_raw = torch.cat([clip_raw, pad_L], dim=2)
clip_raw = clip_raw[:, :, :5] # (1,10,5,384)
cond_clip = clip_raw.unsqueeze(1) # (1,1,10,5,384)
# ------------------------------------------------------------
bucket_raw = last_prompts[:, start:start + 50] # (1, ≤50, 1, 384)
if bucket_raw.shape[1] < 50:
pad_w = torch.zeros_like(bucket_raw[:, :50 - bucket_raw.shape[1]])
bucket_raw = torch.cat([bucket_raw, pad_w], dim=1)
bucket_clip = bucket_raw.unsqueeze(1) # (1,1,50,1,384)
motion = audio2bucket(bucket_clip, image_embeds) * 16 + 16
ref_list.append(ref_img[0])
audio_list.append(audio_pe(cond_clip).squeeze(0)[0])
uncond_list.append(audio_pe(torch.zeros_like(cond_clip)).squeeze(0)[0])
motion_buckets.append(motion[0])
# -------- diffusion --------------------------------------------------
video = pipe(
ref_img, clip_img, face_mask,
audio_list, uncond_list, motion_buckets,
height=height, width=width,
num_frames=len(audio_list),
decode_chunk_size=cfg.decode_chunk_size,
motion_bucket_scale=cfg.motion_bucket_scale,
fps=cfg.fps,
noise_aug_strength=cfg.noise_aug_strength,
min_guidance_scale1=cfg.min_appearance_guidance_scale,
max_guidance_scale1=cfg.max_appearance_guidance_scale,
min_guidance_scale2=cfg.audio_guidance_scale,
max_guidance_scale2=cfg.audio_guidance_scale,
overlap=cfg.overlap,
shift_offset=cfg.shift_offset,
frames_per_batch=cfg.n_sample_frames,
num_inference_steps=cfg.num_inference_steps,
i2i_noise_strength=cfg.i2i_noise_strength,
).frames
video = (video * 0.5 + 0.5).clamp(0, 1)
return video.to(pipe.device).unsqueeze(0).cpu()
# ------------------------------------------------------------------
# Sonic class
# ------------------------------------------------------------------
class Sonic:
config_file = os.path.join(BASE_DIR, "config/inference/sonic.yaml")
config = OmegaConf.load(config_file)
def __init__(self, device_id: int = 0, enable_interpolate_frame: bool = True):
cfg = self.config
cfg.use_interframe = enable_interpolate_frame
self.device = f"cuda:{device_id}" if device_id >= 0 and torch.cuda.is_available() else "cpu"
cfg.pretrained_model_name_or_path = os.path.join(BASE_DIR, cfg.pretrained_model_name_or_path)
self._load_models(cfg)
print("Sonic init done")
# --------------------------------------------------------------
def _load_models(self, cfg):
dtype = {"fp16": torch.float16, "fp32": torch.float32, "bf16": torch.bfloat16}[cfg.weight_dtype]
vae = AutoencoderKLTemporalDecoder.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="vae", variant="fp16")
sched = EulerDiscreteScheduler.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="scheduler")
imgE = CLIPVisionModelWithProjection.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="image_encoder", variant="fp16")
unet = UNetSpatioTemporalConditionModel.from_pretrained(cfg.pretrained_model_name_or_path, subfolder="unet", variant="fp16")
add_ip_adapters(unet, [32], [cfg.ip_audio_scale])
a2t = AudioProjModel(10, 5, 384, 1024, 1024, 32).to(self.device)
a2b = Audio2bucketModel(50, 1, 384, 1024, 1024, 1, 2).to(self.device)
unet.load_state_dict(torch.load(os.path.join(BASE_DIR, cfg.unet_checkpoint_path), map_location="cpu"))
a2t.load_state_dict(torch.load(os.path.join(BASE_DIR, cfg.audio2token_checkpoint_path), map_location="cpu"))
a2b.load_state_dict(torch.load(os.path.join(BASE_DIR, cfg.audio2bucket_checkpoint_path), map_location="cpu"))
whisper = WhisperModel.from_pretrained(os.path.join(BASE_DIR, "checkpoints/whisper-tiny")).to(self.device).eval()
whisper.requires_grad_(False)
self.feature_extractor = AutoFeatureExtractor.from_pretrained(os.path.join(BASE_DIR, "checkpoints/whisper-tiny"))
self.face_det = AlignImage(self.device, det_path=os.path.join(BASE_DIR, "checkpoints/yoloface_v5m.pt"))
if cfg.use_interframe:
self.rife = RIFEModel(device=self.device)
self.rife.load_model(os.path.join(BASE_DIR, "checkpoints/RIFE/"))
for m in (imgE, vae, unet):
m.to(dtype)
self.pipe = SonicPipeline(unet=unet, image_encoder=imgE, vae=vae, scheduler=sched).to(device=self.device, dtype=dtype)
self.image_encoder = imgE
self.audio2token = a2t
self.audio2bucket = a2b
self.whisper = whisper
# --------------------------------------------------------------
def preprocess(self, image_path: str, expand_ratio: float = 1.0):
img = cv2.imread(image_path)
h, w = img.shape[:2]
_, _, bboxes = self.face_det(img, maxface=True)
if bboxes:
x1, y1, ww, hh = bboxes[0]
return {"face_num": 1, "crop_bbox": process_bbox((x1, y1, x1+ww, y1+hh), expand_ratio, h, w)}
return {"face_num": 0, "crop_bbox": None}
# --------------------------------------------------------------
@torch.no_grad()
def process(self, image_path: str, audio_path: str, output_path: str,
min_resolution: int = 512, inference_steps: int = 25,
dynamic_scale: float = 1.0, keep_resolution: bool = False,
seed: int | None = None):
cfg = self.config
if seed is not None:
cfg.seed = seed
cfg.num_inference_steps = inference_steps
cfg.motion_bucket_scale = dynamic_scale
seed_everything(cfg.seed)
test_data = image_audio_to_tensor(
self.face_det, self.feature_extractor,
image_path, audio_path, limit=-1,
image_size=min_resolution, area=cfg.area,
)
if test_data is None:
return -1
h, w = test_data["ref_img"].shape[-2:]
resolution = (f"{(Image.open(image_path).width//2)*2}x{(Image.open(image_path).height//2)*2}"
if keep_resolution else f"{w}x{h}")
video = test(self.pipe, cfg, self.whisper, self.audio2token,
self.audio2bucket, self.image_encoder, w, h, test_data)
if cfg.use_interframe:
out = video.to(self.device)
frames = []
for i in tqdm(range(out.shape[2]-1), ncols=0):
mid = self.rife.inference(out[:,:,i], out[:,:,i+1]).clamp(0,1).detach()
frames.extend([out[:,:,i], mid])
frames.append(out[:,:,-1])
video = torch.stack(frames, 2).cpu()
tmp = output_path.replace(".mp4", "_noaudio.mp4")
save_videos_grid(video, tmp, n_rows=video.shape[0], fps=cfg.fps*(2 if cfg.use_interframe else 1))
os.system(f"ffmpeg -i '{tmp}' -i '{audio_path}' -s {resolution} "
f"-vcodec libx264 -acodec aac -crf 18 -shortest '{output_path}' -y -loglevel error")
os.remove(tmp)
return 0
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